Index: webrtc/api/peerconnectioninterface_unittest.cc |
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc |
index 2c044e20806d1bcf0115f5b263c9640a616e7712..61eb5f0423c4f9a9114eb4140a3ebaaa0ab89155 100644 |
--- a/webrtc/api/peerconnectioninterface_unittest.cc |
+++ b/webrtc/api/peerconnectioninterface_unittest.cc |
@@ -1701,16 +1701,10 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) { |
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state()); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_ReceiveFireFoxOffer DISABLED_ReceiveFireFoxOffer |
-#else |
-#define MAYBE_ReceiveFireFoxOffer ReceiveFireFoxOffer |
-#endif |
// Test that we can create a session description from an SDP string from |
// FireFox, use it as a remote session description, generate an answer and use |
// the answer as a local description. |
-TEST_F(PeerConnectionInterfaceTest, MAYBE_ReceiveFireFoxOffer) { |
+TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) { |
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
@@ -2094,19 +2088,11 @@ TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) { |
EXPECT_EQ(0u, observer_.remote_streams()->count()); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_SdpWithoutMsidCreatesDefaultStream \ |
- DISABLED_SdpWithoutMsidCreatesDefaultStream |
-#else |
-#define MAYBE_SdpWithoutMsidCreatesDefaultStream \ |
- SdpWithoutMsidCreatesDefaultStream |
-#endif |
// This tests that a default MediaStream is created if a remote session |
// description doesn't contain any streams and no MSID support. |
// It also tests that the default stream is updated if a video m-line is added |
// in a subsequent session description. |
-TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) { |
+TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2132,18 +2118,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) { |
remote_stream->GetVideoTracks()[0]->state()); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \ |
- DISABLED_SendOnlySdpWithoutMsidCreatesDefaultStream |
-#else |
-#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \ |
- SendOnlySdpWithoutMsidCreatesDefaultStream |
-#endif |
// This tests that a default MediaStream is created if a remote session |
// description doesn't contain any streams and media direction is send only. |
TEST_F(PeerConnectionInterfaceTest, |
- MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream) { |
+ SendOnlySdpWithoutMsidCreatesDefaultStream) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2175,19 +2153,11 @@ TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) { |
// No crash is a pass. |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \ |
- DISABLED_SdpWithoutMsidAndStreamsCreatesDefaultStream |
-#else |
-#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \ |
- SdpWithoutMsidAndStreamsCreatesDefaultStream |
-#endif |
// This tests that a default MediaStream is created if the remote session |
// description doesn't contain any streams and don't contain an indication if |
// MSID is supported. |
TEST_F(PeerConnectionInterfaceTest, |
- MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream) { |
+ SdpWithoutMsidAndStreamsCreatesDefaultStream) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2200,17 +2170,9 @@ TEST_F(PeerConnectionInterfaceTest, |
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size()); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_SdpWithMsidDontCreatesDefaultStream \ |
- DISABLED_SdpWithMsidDontCreatesDefaultStream |
-#else |
-#define MAYBE_SdpWithMsidDontCreatesDefaultStream \ |
- SdpWithMsidDontCreatesDefaultStream |
-#endif |
// This tests that a default MediaStream is not created if the remote session |
// description doesn't contain any streams but does support MSID. |
-TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) { |
+TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2219,19 +2181,11 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) { |
EXPECT_EQ(0u, observer_.remote_streams()->count()); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_DefaultTracksNotDestroyedAndRecreated \ |
- DISABLED_DefaultTracksNotDestroyedAndRecreated |
-#else |
-#define MAYBE_DefaultTracksNotDestroyedAndRecreated \ |
- DefaultTracksNotDestroyedAndRecreated |
-#endif |
// This tests that when setting a new description, the old default tracks are |
// not destroyed and recreated. |
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250 |
TEST_F(PeerConnectionInterfaceTest, |
- MAYBE_DefaultTracksNotDestroyedAndRecreated) { |
+ DefaultTracksNotDestroyedAndRecreated) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2266,17 +2220,11 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) { |
EXPECT_EQ(0u, observer_.remote_streams()->count()); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_LocalDescriptionChanged DISABLED_LocalDescriptionChanged |
-#else |
-#define MAYBE_LocalDescriptionChanged LocalDescriptionChanged |
-#endif |
// This tests that an RtpSender is created when the local description is set |
// after adding a local stream. |
// TODO(deadbeef): This test and the one below it need to be updated when |
// an RtpSender's lifetime isn't determined by when a local description is set. |
-TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) { |
+TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2312,18 +2260,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) { |
EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1])); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \ |
- DISABLED_AddLocalStreamAfterLocalDescriptionChanged |
-#else |
-#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \ |
- AddLocalStreamAfterLocalDescriptionChanged |
-#endif |
// This tests that an RtpSender is created when the local description is set |
// before adding a local stream. |
TEST_F(PeerConnectionInterfaceTest, |
- MAYBE_AddLocalStreamAfterLocalDescriptionChanged) { |
+ AddLocalStreamAfterLocalDescriptionChanged) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2349,18 +2289,10 @@ TEST_F(PeerConnectionInterfaceTest, |
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1])); |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \ |
- DISABLED_ChangeSsrcOnTrackInLocalSessionDescription |
-#else |
-#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \ |
- ChangeSsrcOnTrackInLocalSessionDescription |
-#endif |
// This tests that the expected behavior occurs if the SSRC on a local track is |
// changed when SetLocalDescription is called. |
TEST_F(PeerConnectionInterfaceTest, |
- MAYBE_ChangeSsrcOnTrackInLocalSessionDescription) { |
+ ChangeSsrcOnTrackInLocalSessionDescription) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |
@@ -2404,18 +2336,10 @@ TEST_F(PeerConnectionInterfaceTest, |
// changed. |
} |
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
-#if defined(WEBRTC_WIN) && defined(_DEBUG) |
-#define MAYBE_SignalSameTracksInSeparateMediaStream \ |
- DISABLED_SignalSameTracksInSeparateMediaStream |
-#else |
-#define MAYBE_SignalSameTracksInSeparateMediaStream \ |
- SignalSameTracksInSeparateMediaStream |
-#endif |
// This tests that the expected behavior occurs if a new session description is |
// set with the same tracks, but on a different MediaStream. |
TEST_F(PeerConnectionInterfaceTest, |
- MAYBE_SignalSameTracksInSeparateMediaStream) { |
+ SignalSameTracksInSeparateMediaStream) { |
FakeConstraints constraints; |
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
true); |